Thanks Joe for your prompt,
For interest, which Asterisk version.
I'm using asterisk 184.108.40.206& a2billing 1.9.4
Are you using realtime?
Yes sure, I'm using realtime, for that reason it's using the DB.
Has the customer made a call, or will it just change by itself?
I guess it's changing when the customer try to register, NOT when it's try to make a call, cause only registered customers are being modified... I removed my cronjobs in the system, but no way, it's the same, in both servers, sip user when try to register it's being modified to "s" in the database, then the user it's registered, as Name/username= 764570/s, then it cann't place calls...in cli, when the user try to call, i see this:
[Jul 5 10:44:40] WARNING: channel.c:5635 ast_channel_make_compatible_helper: No path to translate from SIP/TrunkC-00000029 to SIP/27567-00000028
-- AGI Script Executing Application: (DIAL) Options: (SIP/TrunkD/00656476688,60,LIW(13560000:60000:30000))
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called TrunkD/00656476688
[Jul 5 10:44:40] WARNING: channel.c:5635 ast_channel_make_compatible_helper: No path to translate from SIP/TrunkD-0000002a to SIP/27567-00000028
-- <SIP/27567-00000028>AGI Script a2billing.php completed, returning -1
The other question, when I edit the customer voip account, and change it's host to static, from dynamic, the username is not being altered any more, I understand with that the sip user is being registered just with the IP, before the invite for username, so it's being modified exactly in the registrar time...
Thanks in advance,